--- (10 headers 0 lines) --- Really destroying SIP dialog '1ef6d40547e0100e6acc8c9d584a91f0@xxx.xxx.xxx.xxx' Method: OPTIONS Reliably Transmitting (NAT) to 10.1.50.60:5060: OPTIONS sip:sipuser16@10.1.50.60 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3f7592c0;rport From: "asterisk" ;tag=as6ce753cd To: Contact: Call-ID: 33328017588542f818c1ff0c4f586a6b@xxx.xxx.xxx.xxx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 09 Feb 2008 22:44:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- vp0*CLI> no sip debug <--- SIP read from 10.1.50.60:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3f7592c0;received=xxx.xxx.xxx.xxx;rport=5060 From: "asterisk" ;tag=as6ce753cd To: ;tag=as36cf0a57 Call-ID: 33328017588542f818c1ff0c4f586a6b@xxx.xxx.xxx.xxx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0